conncetion asterisk from outside network via sip - networking

I have asterisk in a server having public ip. I am trying to asterisk from outside network from a sip phone(zoipar). I have opened the port 5060 on my router which is the default udp port for asterisk sip connection and i have also opened the 10000-20000 port for rtp defined in rtp.conf in asterisk.
When i m trying to connect my softphone to asterisk server from outside my network, it says Registration timeout and when i check if i got any hit on my port 5060, its doesnt show anything.
on my server 5060 is running
netstat -nlp | grep 5060
udp 0 0 0.0.0.0:5060 0.0.0.0:* 21768/asterisk
BTW I'm able to connect from local network without any problem .

You need to forward incoming traffic on your router from SIP and RTP to your asterisk server, it's not enough to open those ports, you need to explain your router where to send incoming traffic ton those ports

You need setup NAT.
This article will help you in your situation.
http://www.voip-info.org/wiki/view/Asterisk+sip+nat

You need to login to your router and forward the ports to your asterisk server internal IP.
You will also need to make sure your firewall on your server is setup correctly to allow the ports to go in and out of your server.
You can read more on iptables here: http://www.cyberciti.biz/tips/linux-iptables-examples.html

Related

TCP packet transfer when sender and receiver are same system

I am running server and client in same system. TCP protocol is used for communication between them. In a scenario where client wants to send packet to server, will it go through network infra (i.e. router, internet etc) and come to server or will it manage transfer within system and ignore network.
If you are on the server, any communication you initiate to IP addresses also on the same server will never leave the server.
You can test by installing tcpdump then running from the console/keyboard/mouse:
tcpdump -n -i enp0s5 not arp
Do not generate network traffic. Try to ssh to your account on IP 127.0.0.1 (e.g. risner#127.0.0.1).
Also try to initiate ssh to another host on the network.
Nothing should show on tcpdump, so that indicates it is not leaving the machine.

How to create TCP tunnels with Pagekite

I am a complete beginner when it comes to networking and I am trying to set up a TCP tunnel on my machine using pagekite. I want to route all traffic from a TCP address to a port on my localhost, let's say 8080. I would then start a handler on localhost:8080 to deal with the incoming traffic. I can get this to work with ngrok simply by doing ngrok tcp 8080, but on a free ngrok plan I cannot reserve tcp addresses and ngrok is rather slow, so I opted to try and use pagekite.
Pagekite normally allows easy tunnelling to an HTTP address, but they have a guide here about how to use PuTTY along with Pagekite to create a TCP tunnel proxied by HTTP.
I followed their guide but could use some help figuring out if it does what I want it to do.
I am working on a Linux VM, so I first set up an SSH server with openssh like this: sudo service ssh start
I then exposed that SSH server using pagekite like this: python3 pagekite.py 22 ssh:user.pagekite.me
I then started PuTTY, and configured the Host Name to be user.pagekite.me on port 22, setup an HTTP proxy with the proxy hostname user.pagekite.me on port 443 and finally created a tunnel from the PuTTY machine with source port 8080 and destination localhost:8080.
Now I am not sure what this actually accomplished. I know that the PuTTY machine connected to the ssh server running on my VM and I am able to use the linux terminal from the PuTTY terminal but has this actually created a TCP tunnel from user.pagekite.me:8080 to localhost:8080? Additionally after doing this, if I try to setup the handler on localhost:8080 I get the following error:
Handler failed to bind to 0.0.0.0:8080
Rex::BindFailed The address is already in use or unavailable: (0.0.0.0:8080).
Again I am completely clueless when it comes to networking so if anyone could explain what it is I'm doing and if it is even possible to do what I want the way that I am doing it, that would be quite helpful.

How to enable port for public access?

I have enabled 1 port [8081] and it's accessible from the remote computer. but the same for other port [7500] not working?
I would like to know the meaning of the below line?
TCP [::]:8081 [::]:0 LISTENING
And how to enable the same for port [7500]?
Attached listening port status:
netstat -na outputs 4 columns of data:
Proto, Local Address, Foreign Address, and State.
When looking for port 8081, you find 2 entries - one for TCP on 0.0.0.0:8081 for IPv4, and one for TCP [::]:8081 for IPv6.
When looking for port 7500, you find 1 entry - one for TCP 0.0.0.0:7500 for IPv4 only.
In both cases, you have local sockets listening via wildcard IPs to all local network adapters, and there is no "Foriegn Address" assigned because a listening socket is not connected to any remote party. TCP sockets in the ESTABLISHED state have remote parties.
You have not shown any code, or explained your network setup, so nobody can really explain why you have 2 entries for port 8081 but only 1 entry for port 7500, or why remote computers can connect to port 8081 but not to port 7500. Maybe those clients are only using IPv6? Maybe your listening computer is behind a router that doesn't forward port 7500? We don't know.

NAT configuration for SIP

I have this network configuration:
SIP Provider -> Mikrotik CCR as my network Gateway (NAT) -> Asterisk PBX -> Mikrotik RB as CPE device (NAT) -> SIP Devices behind the RB.
How I have to configure the NAT on asterisk and on Customer devices?
Actually I have some problems of one-way-audio but not on all calls.
Thank you in advice for the help
Asterisk can both act as a SIP client and a SIP server. Asterisk as a SIP client is configured with type=peer (or type=friend) in one or more client sections of sip.conf and, optionally, one or more register=> lines in the [general] section of sip.conf. Asterisk as a SIP server connects clients (SIP Phones) configured by specifying their own username, secret, etc. (and either type=peer or type=friend) in client sections of sip.conf.
Asterisk SIP channels in a NATed network can be generalized like this:
Asterisk as a SIP client behind nat, connecting to outside SIP Proxies
Asterisk as a SIP client behind nat, connecting to inside SIP proxies
Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk
Asterisk as a SIP server behind nat, clients on the outside behind a second NAT connecting to Asterisk
Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk
Asterisk as a SIP client outside nat, connecting to outside SIP proxies
Asterisk as a SIP client outside nat, connecting to inside SIP proxies
Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk
Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk
Every setup works somewhere, but it depends on the client, the NAT, the server and many other factors. In most cases, 1 and 3 is broken. SIP is a peer-to-peer protocol and a NAT can be generalized and simplified as a solution that allows clients on the inside to connect to servers on the outside and _not_ allow clients on the outside to connect to any server on the inside.
#1 works with a NAT-supporting proxy as SIP Express router as the outside proxy. (Get an account at IPtel.org and try!). Fails with Free World Dialup.
#2 Works- no NAT in between
#3 Works with port forwarding and some header mangling magic tips
#4 Works with port forwarding, STUN on the remote and some fine tuning of RTP port allocation
#5 Works - no NAT in between
#6 is no problem. No NAT in the middle
#7 is a problem if no port forwarding is done, similar to 3 above.
#8 is no problem. No NAT in the middle
#9 is solved with nat=yes and qualify=xxx in sip.conf for the client in most cases. Some clients (X-lite) assist themselves by using STUN and sending UDP keep-alive packets. Qualify sends keep-alive packets from Asterisk to the client on the inside.
http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

What happens when 2 computers listen to the same port and a router receives a packet through that port

What I am asking is if two computers listen to the same port and a packet of information enters the router through the WAN Ip and the same port. Would the packet go to both computers? Neither? One or the other?
IE
computer 1 -(internal IP)-> 192.168.1.3 -(listens to port)-> 4444
computer 2 -(internal IP)-> 192.168.1.2 -(listens to port)-> 4444
computer 3 -(connects and sends)-> 24.157.358.45:4444
packet -> computer 1 AND computer 2
The code in VB6 is:
LAN.LocalPort = 4444
LAN.Protocol = sckTCPProtocol
LAN.Listen
I am using a WinSock object in the Microsoft WinSock Control 6.0 in VB6 Professional
If there is something that needs to be clarified I would be more than happy to.
The router won't send an inbound packet to either machine unless communication has already been established.
If 192.168.1.3 calls out to some other machine (e.g. 4.5.6.7) from its port 4444, the router will assign an arbitrary port on its external address (say 24.157.358.45 [sic] :5555) and pass the packets on to 4.5.6.7. 4.5.6.7 will send reply packets to 24.157.358.45:5555 -- because that's the only address it knows about -- and the router will relay those to 192.168.1.3:4444.
That's the normal course of things, but there are a lot of additional details to this scheme that make it possible to establish communication with a machine behind a router via trickery.
The system of having machines with private IP addresses behind a router with a public address is called network address translation (NAT); it's a pretty deep topic.
From my knowledge of routers, unless port forwarding is setup, the router will discard any packets sent on that port.
If port forwarding is setup, only one of the computers could be setup to receive the packets.
If the packet is an inbound request to establish a new TCP connection with a server that is running behind the router, the router must have an explicit port-forwarding rule configured, either statically in the router's configuration or dynamically via uPNP or SNMP, that tells the router where to route inbound packets on 24.157.358.45:4444 to, either to 192.168.1.2:4444 or to 192.168.1.3:4444, otherwise the packet will be discarded. So no, both of your listening servers will not see the same packet.
Once a TCP connection is established, the router knows which specific LAN machines are associated with which connections and will route incoming packets belonging to those connections accordingly.
The previous answers are correct, you need to enable port forwarding. If it is not enabled port 4444 will be closed on the router.
It sounds like you have multiple servers and want to forward to whatever server is turned on at the moment. This is not possible (*), the router does not care whether or not PC1 or PC2 are listening on port 4444, it will simply forward everything to the address configured in the port forwarding.
(*): Ok it is possible but it takes some extra work.
Solution 1: Trick the router into thinking there is only one server. Give PC1 and PC2 a virtual network interface with the same IP address and forward to that address. Make sure only one of these interfaces is enabled, having duplicate IP addresses in your network can have unintentional behaviour.
Solution 2: Make the router care about which server is on. You will need to write a program to run on the router (or on another server) that can detect which server is on and forward the packets accordingly. If you are using Linux the program iptables can be worth looking at.

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